libavformat/audiointerleave.c

   1 /*
   2  * Audio Interleaving functions
   3  *
   4  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 #include "libavutil/fifo.h"
  24 #include "avformat.h"
  25 #include "audiointerleave.h"
  26 #include "internal.h"
  27
  28 void ff_audio_interleave_close(AVFormatContext *s)
  29 {
  30     int i;
  31     for (i = 0; i < s->nb_streams; i++) {
  32         AVStream *st = s->streams[i];
  33         AudioInterleaveContext *aic = st->priv_data;
  34
  35         if (st->codec->codec_type == CODEC_TYPE_AUDIO)
  36             av_fifo_free(aic->fifo);
  37     }
  38 }
  39
  40 int ff_audio_interleave_init(AVFormatContext *s,
  41                              const int *samples_per_frame,
  42                              AVRational time_base)
  43 {
  44     int i;
  45
  46     if (!samples_per_frame)
  47         return -1;
  48
  49     for (i = 0; i < s->nb_streams; i++) {
  50         AVStream *st = s->streams[i];
  51         AudioInterleaveContext *aic = st->priv_data;
  52
  53         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
  54             aic->sample_size = (st->codec->channels *
  55                                 av_get_bits_per_sample(st->codec->codec_id)) / 8;
  56             if (!aic->sample_size) {
  57                 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
  58                 return -1;
  59             }
  60             aic->samples_per_frame = samples_per_frame;
  61             aic->samples = aic->samples_per_frame;
  62             aic->time_base = time_base;
  63
  64             aic->fifo_size = 100* *aic->samples;
  65             aic->fifo= av_fifo_alloc(100 * *aic->samples);
  66         }
  67     }
  68
  69     return 0;
  70 }
  71
  72 static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
  73                                    int stream_index, int flush)
  74 {
  75     AVStream *st = s->streams[stream_index];
  76     AudioInterleaveContext *aic = st->priv_data;
  77
  78     int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
  79     if (!size || (!flush && size == av_fifo_size(aic->fifo)))
  80         return 0;
  81
  82     av_new_packet(pkt, size);
  83     av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
  84
  85     pkt->dts = pkt->pts = aic->dts;
  86     pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
  87     pkt->stream_index = stream_index;
  88     aic->dts += pkt->duration;
  89
  90     aic->samples++;
  91     if (!*aic->samples)
  92         aic->samples = aic->samples_per_frame;
  93
  94     return size;
  95 }
  96
  97 int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
  98                         int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
  99                         int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
 100 {
 101     int i;
 102
 103     if (pkt) {
 104         AVStream *st = s->streams[pkt->stream_index];
 105         AudioInterleaveContext *aic = st->priv_data;
 106         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
 107             unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
 108             if (new_size > aic->fifo_size) {
 109                 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
 110                     return -1;
 111                 aic->fifo_size = new_size;
 112             }
 113             av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
 114         } else {
 115             // rewrite pts and dts to be decoded time line position
 116             pkt->pts = pkt->dts = aic->dts;
 117             aic->dts += pkt->duration;
 118             ff_interleave_add_packet(s, pkt, compare_ts);
 119         }
 120         pkt = NULL;
 121     }
 122
 123     for (i = 0; i < s->nb_streams; i++) {
 124         AVStream *st = s->streams[i];
 125         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
 126             AVPacket new_pkt;
 127             while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
 128                 ff_interleave_add_packet(s, &new_pkt, compare_ts);
 129         }
 130     }
 131
 132     return get_packet(s, out, pkt, flush);
 133 }